As provision of voice telephone services on a world wide basis progresses, an expanding variety of networks are being utilized to handle the high volume of traffic throughput. The recent development of data network capability, such as the Internet, for voice call communication offers advantages that are now being explored. The Internet was initially intended to accommodate transmission of data, not necessarily on a real time basis. Acclimation of such data networks to the stringent voice transmission requirements of real time, high quality communication presents developmental challenges.
The emergence of Integrated Services Digital Network (ISDN) technology has enabled local area networks to be interconnected with each other to form a wider area network and to enable a local area network to be accessed by a remote personal computer and operated as if the computer were resident on the network. ISDN has integrated computer and communications technologies to provide, worldwide, a common, all-digital network. By virtue of a standardized structure of digital protocols, implementation of multiple networks within national boundaries, appears to a user as a single, uniformly accessible, worldwide network capable of handling a broad range of telephone, data and other conventional and enhanced services.
ISDN is configured for carrying both voice and data communication. Within the framework of ISDN, the network provides services and the user accesses the services through the user-network interface. A "channel" represents a specified portion of the information carrying capacity of an interface. Channels are classified by two types, Basic Rate ISDN (BRI) and Primary Rate ISDN (PRI). BRI delivers two B-channels, each having a capacity of 64 Kbps, capable of transmitting voice and data simultaneously. A 16 Kbps D-channel transmits call control messages and user packet data. PRI provides twenty three B-channels of 64 Kbps capacity each for carrying voice, circuit switched data or packet data. The D-channel is a 64 Kbps signaling channel. The B and D channels are logically multiplexed together. Particular description of conventional ISDN interfaces at the customer premises, the local loop at the carrier end and exchange switching equipment is not believed necessary to the present disclosure. Details of such architecture may be found in ISDN: An Overview, Data Pro Research, Concepts & Technologies, MT 20-365; pp 101-110, published by McGraw Hill, Incorporated (December 1988).
The Internet basically comprises several large computer networks joined together over high-speed data links ranging from ISDN to T1, T3, FDDI, SONET, SMDS, OT1, etc. The most prominent of these national nets are MILNET (Military Network), NSFNET (National Science Foundation NETwork), and CREN (Corporation for Research and Educational Networking). Network standards, conventions and protocols have evolved for interconnecting networks and routing information in an orderly manner. These protocols, commonly referred to as TCP/IP (Transport Control Protocol/Internet Protocol), have become widely used in the industry. TCP/IP is flexible and robust. In effect, TCP takes care of the integrity and IP moves the data. Internet provides two broad types of data services: connectionless packet delivery service and reliable stream transport service. To accommodate telephone service for usage by ordinary analog telephone sets, analog voice signals are converted to appropriate data format for transmission through data networks and then reconverted to analog before being received at the destination.
A simplified diagram of the Internet is depicted in FIG. 1. The Internet 50 comprises Autonomous Systems (AS) which may be owned and operated by Internet Service Providers (ISPs) such as PSI, UUNET, MCI, SPRINT, etc. Three such AS/ISPs are shown in FIG. 1 at 52, 54 and 56. The Autonomous Systems are linked by Inter-AS Connections 58, 60 and 62. Information Providers (IPs) 64 and 66, such as America Online (AOL) and Compuserve, are connected to the Internet via high speed lines 68 and 70, such as T1/T3 and the like. Information Providers generally do not have their own Internet based Autonomous Systems but have or use Dial-Up Networks such as SprintNet (X.25), DATAPAC and TYMNET.
Other information providers, such as universities, are indicated in exemplary fashion at 72 and are connected to the AS/ISPs via the same type connections, here illustrated as T1 lines 74. Corporate Local Area Networks (LANs), such as those illustrated at 76 and 78, are connected to the Internet through routers 80 and 82 and links shown as T1 lines 84 and 86. Laptop or PC computers 88 and 90 are representative of computers connected to the Internet via the public switched telephone network (PSTN), shown connected to the AS/ISPs via dial up links 92 and 96.
The Information Providers (IPs) are end systems that collect and market the information through their own servers. Access providers are companies such as UUNET, PSI, MCI and SPRINT which transport the information. Such companies market the usage of their networks.
In simplified fashion the Internet may be viewed as a series of gateway routers connected together with computers connected to the routers. In the addressing scheme of the Internet an address comprises four numbers separated by dots. An example would be 164.109.211.237. Each machine on the Internet has a unique number that includes one of these four numbers. In the address, the leftmost number is the highest number. By analogy this would correspond to the ZIP code in a mailing address. The first two numbers that constitute this portion of the address may indicate a network or a locale. That network is connected to the last router in the transport path. In differentiating between two computers in the same destination network only the last number field changes. In such an example the next number field 211 identifies the destination router. When the packet bearing the destination address leaves the source router it examines the first two numbers in a matrix table to determine how many hops are the minimum to get to the destination. It then sends the packet to the next router as determined from that table and the procedure is repeated. Each router has a database table that finds the information automatically. This process continues until the packet arrives at the destination computer. The separate packets that constitute a message may not travel the same path, depending on traffic load. However, they all reach the same destination and are assembled in their original sequence order in a connectionless fashion. This is in contrast to connection oriented modes such as frame relay and ATM or voice.
Software has recently been developed for use on personal computers to permit two-way transfer of real-time voice information via an Internet data link between two personal computers. In one of the directions, the sending computer converts voice signals from analog to digital format. The software facilitates data compression down to a rate compatible with modem communication via a POTS telephone line. The software also facilitates encapsulation of the digitized and compressed voice data into the TCP/IP protocol, with appropriate addressing to permit communication via the Internet. At the receiving end, the computer and software reverse the process to recover the analog voice information for presentation to the other party. Such programs permit telephone-like communication between Internet users registered with Internet Phone Servers. The book "Mastering the Internet", Glee Cady and Pat McGregor, SYBEX Inc., Alameda, Calif., 1994, ISBN 94-69309, very briefly describes three proprietary programs said to provide real-time video and voice communications via the Internet.
The commonly assigned applications, Ser. Nos. 08/634,543 and 08/670,908, identified more particularly above, are concerned with providing telephone service via the Internet to users of the public telecommunications network who may not have access to a computer or other access to the Internet. Conversion of analog voice signals into digital format appropriate for Internet transmission and conversion of digital signals received from the Internet back to voice analog signals are functions performed by the telephone service provider rather than a POTS subscriber's PC. Such service would be economical, especially for long distance calls, compared with the toll rates charged by long distance interexchange carriers.
With increasing volume of use on the Internet and the bursty nature of data transmission, traffic patterns have become unstable and unpredictable. The minimum quality of service acceptable for voice communication is much higher than the level for data transport as transmission delays noticeably degrade conversation. With the Internet or other high volume data network, acceptable voice communication may be available between two end points at one given time, but not at other times. A surge in data traffic may make the network unsuitable for voice communication for as much as twenty or thirty minutes. Bottlenecks may occur at different points in the network at different times. The locations of the participants of a voice call are factors in determining suitability of the data network. The degree to which degradation of a voice call remains acceptable is subjective with the user and can be a tradeoff between quality of service and reduction of cost.
A deficiency in earlier proposed voice Internet service systems is the inability to ensure an acceptable level of service quality. Voice communication by nature should be perceived as real time interaction in order to be acceptable to the parties to the call. The packet data network traffic in the connection paths of a voice call may render intolerable transmission delays. A high level of congestion and delay in a data network often leads to lost or dropped data packets that would noticeably degrade reconstructed voice audio. Current systems do not measure delays against user acceptable standards. The voice call user must either endure such deficiencies or terminate the call in favor of originating a new call through an alternative system.
The aforementioned commonly assigned application Ser. No. 08/821,027 (Attorney docket No. 680-189), is concerned with determining routing of voice calls alternatively between the public switched telephone network (PSTN) and a data packet network, such as the Internet, in accordance with the quality of service existing in the data packet network at the times of call origination. Through use of the PSTN Advanced Intelligent Network (AIN), a caller may predefine an acceptable level of service, for example 2.4 or 4.8 kbs to be stored in the user's Call Processing Record (CPR) in the AIN Integrated Services Control Point (ISCP). On a per call basis, the caller linked to a first public switched network may indicate a preference to route through the Internet. This indication would be recognized by the AIN system, in response to which the quality of service currently present on the Internet for completion of the call is measured. If the result exceeds the stored threshold, the call is set up and routed through the Internet to the switched network link to the destination party. If the quality of service on the Internet is not satisfactory, the call would be alternatively routed through the PSTN, which may include an Interexchange Carrier link.
The last described arrangement is an improvement over prior voice data network schemes in the respect that determination of data network performance quality avoids set up of a call that can be known at the outset to be inadequate for voice communication. However, with relatively unstable and unpredictable traffic patterns in data networks such as the Internet, the alternative set up arrangement does not accommodate for degradation of data network performance conditions after a call has been placed and routed through the data network. Thus, parties to such a call still must either suffer the deficiencies in voice quality, perhaps in the hope that data traffic conditions improve, or terminate the call in favor of a new call manually placed through the switched telephone network.
The aforementioned commonly assigned application Ser. No. 08/815,361 (Attorney docket No. 680-190), is concerned with monitoring the quality of service existing in a data packet network during the course of communication of a voice call through the data network. The user's acceptable level of service may be predefined with a threshold quality level stored in the user's Call Processing Record (CPR) in the AIN Integrated Services Control Point (ISCP). If the monitored quality is maintained in excess of the stored threshold, communication of the call continues through the established course of transmission. If the measured quality of service on the data network is not satisfactory, the routing of the call is changed to communication solely through a traditional PSTN voice telephone network connection, which may include an Interexchange Carrier link, without terminating the call. The packet data network is bypassed to obtain voice grade quality while maintaining the call between the parties. The call through the data packet network is terminated at that time.
While these quality oriented schemes significantly improve so-called Internet type voice call service by avoiding transmission of calls of unacceptable quality through the data network, they have not eliminated remaining drawbacks. Monitoring data network conditions between two remotely located end servers on an individual call basis entails a redundancy of operation which becomes considerable with a high volume of calls. Changing the routing of an Internet voice call to a traditional POTS call through the PSTN negates the benefits of Internet transmission, such as economies of cost and transmission efficiency, for the remainder of the call. A voice call that has been diverted to the PSTN because of transient high network traffic volume is not rerouted back through the data network when data network conditions again become favorable.